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      1 /*
      2  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
      3  *
      4  *  Use of this source code is governed by a BSD-style license
      5  *  that can be found in the LICENSE file in the root of the source
      6  *  tree. An additional intellectual property rights grant can be found
      7  *  in the file PATENTS.  All contributing project authors may
      8  *  be found in the AUTHORS file in the root of the source tree.
      9  */
     10 
     11 #if defined(WEBRTC_ANDROID)
     12 #include "webrtc/modules/audio_device/android/audio_device_template.h"
     13 #include "webrtc/modules/audio_device/android/audio_record_jni.h"
     14 #include "webrtc/modules/audio_device/android/audio_track_jni.h"
     15 #if !defined(WEBRTC_CHROMIUM_BUILD)
     16 #include "webrtc/modules/audio_device/android/opensles_input.h"
     17 #include "webrtc/modules/audio_device/android/opensles_output.h"
     18 #endif
     19 #endif
     20 
     21 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
     22 #include "webrtc/system_wrappers/interface/trace.h"
     23 #include "webrtc/voice_engine/voice_engine_impl.h"
     24 
     25 namespace webrtc
     26 {
     27 
     28 // Counter to be ensure that we can add a correct ID in all static trace
     29 // methods. It is not the nicest solution, especially not since we already
     30 // have a counter in VoEBaseImpl. In other words, there is room for
     31 // improvement here.
     32 static int32_t gVoiceEngineInstanceCounter = 0;
     33 
     34 VoiceEngine* GetVoiceEngine(const Config* config, bool owns_config)
     35 {
     36 #if (defined _WIN32)
     37   HMODULE hmod = LoadLibrary(TEXT("VoiceEngineTestingDynamic.dll"));
     38 
     39   if (hmod) {
     40     typedef VoiceEngine* (*PfnGetVoiceEngine)(void);
     41     PfnGetVoiceEngine pfn = (PfnGetVoiceEngine)GetProcAddress(
     42         hmod,"GetVoiceEngine");
     43     if (pfn) {
     44       VoiceEngine* self = pfn();
     45       if (owns_config) {
     46         delete config;
     47       }
     48       return (self);
     49     }
     50   }
     51 #endif
     52 
     53     VoiceEngineImpl* self = new VoiceEngineImpl(config, owns_config);
     54     if (self != NULL)
     55     {
     56         self->AddRef();  // First reference.  Released in VoiceEngine::Delete.
     57         gVoiceEngineInstanceCounter++;
     58     }
     59     return self;
     60 }
     61 
     62 int VoiceEngineImpl::AddRef() {
     63   return ++_ref_count;
     64 }
     65 
     66 // This implements the Release() method for all the inherited interfaces.
     67 int VoiceEngineImpl::Release() {
     68   int new_ref = --_ref_count;
     69   assert(new_ref >= 0);
     70   if (new_ref == 0) {
     71     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1,
     72                  "VoiceEngineImpl self deleting (voiceEngine=0x%p)",
     73                  this);
     74 
     75     // Clear any pointers before starting destruction. Otherwise worker-
     76     // threads will still have pointers to a partially destructed object.
     77     // Example: AudioDeviceBuffer::RequestPlayoutData() can access a
     78     // partially deconstructed |_ptrCbAudioTransport| during destruction
     79     // if we don't call Terminate here.
     80     Terminate();
     81     delete this;
     82   }
     83 
     84   return new_ref;
     85 }
     86 
     87 VoiceEngine* VoiceEngine::Create() {
     88   Config* config = new Config();
     89   return GetVoiceEngine(config, true);
     90 }
     91 
     92 VoiceEngine* VoiceEngine::Create(const Config& config) {
     93   return GetVoiceEngine(&config, false);
     94 }
     95 
     96 int VoiceEngine::SetTraceFilter(unsigned int filter)
     97 {
     98     WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
     99                  VoEId(gVoiceEngineInstanceCounter, -1),
    100                  "SetTraceFilter(filter=0x%x)", filter);
    101 
    102     // Remember old filter
    103     uint32_t oldFilter = Trace::level_filter();
    104     Trace::set_level_filter(filter);
    105 
    106     // If previous log was ignored, log again after changing filter
    107     if (kTraceNone == oldFilter)
    108     {
    109         WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1,
    110                      "SetTraceFilter(filter=0x%x)", filter);
    111     }
    112 
    113     return 0;
    114 }
    115 
    116 int VoiceEngine::SetTraceFile(const char* fileNameUTF8,
    117                               bool addFileCounter)
    118 {
    119     int ret = Trace::SetTraceFile(fileNameUTF8, addFileCounter);
    120     WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
    121                  VoEId(gVoiceEngineInstanceCounter, -1),
    122                  "SetTraceFile(fileNameUTF8=%s, addFileCounter=%d)",
    123                  fileNameUTF8, addFileCounter);
    124     return (ret);
    125 }
    126 
    127 int VoiceEngine::SetTraceCallback(TraceCallback* callback)
    128 {
    129     WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
    130                  VoEId(gVoiceEngineInstanceCounter, -1),
    131                  "SetTraceCallback(callback=0x%x)", callback);
    132     return (Trace::SetTraceCallback(callback));
    133 }
    134 
    135 bool VoiceEngine::Delete(VoiceEngine*& voiceEngine)
    136 {
    137     if (voiceEngine == NULL)
    138         return false;
    139 
    140     VoiceEngineImpl* s = static_cast<VoiceEngineImpl*>(voiceEngine);
    141     // Release the reference that was added in GetVoiceEngine.
    142     int ref = s->Release();
    143     voiceEngine = NULL;
    144 
    145     if (ref != 0) {
    146         WEBRTC_TRACE(kTraceWarning, kTraceVoice, -1,
    147             "VoiceEngine::Delete did not release the very last reference.  "
    148             "%d references remain.", ref);
    149     }
    150 
    151     return true;
    152 }
    153 
    154 #if !defined(WEBRTC_CHROMIUM_BUILD)
    155 int VoiceEngine::SetAndroidObjects(void* javaVM, void* env, void* context)
    156 {
    157 #ifdef WEBRTC_ANDROID
    158 #ifdef WEBRTC_ANDROID_OPENSLES
    159   typedef AudioDeviceTemplate<OpenSlesInput, OpenSlesOutput>
    160       AudioDeviceInstance;
    161 #else
    162   typedef AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>
    163       AudioDeviceInstance;
    164 #endif
    165   if (javaVM && env && context) {
    166     AudioDeviceInstance::SetAndroidAudioDeviceObjects(javaVM, env, context);
    167   } else {
    168     AudioDeviceInstance::ClearAndroidAudioDeviceObjects();
    169   }
    170   return 0;
    171 #else
    172   return -1;
    173 #endif
    174 }
    175 #endif
    176 
    177 }  // namespace webrtc
    178